I really can't hear much differance between 128kbps ACC files and 160kbps mp3 flies, but i haven't made the jump over to ACC yet and the extra quality on my mp3s gives me piece of mind (even if i can't hear the differance)

I think i will switch over to ACC soon, most likely 160kbps or higher.

I used to encode with iTunes at 128 kbits, but I realized when listening through hifi equipment the quality wasn't that good. After some researching and many comparisons I have to say that I get by far the best results by using LAME and encoding using the standard setting which produces VBR files. The average bit rate is around 180 kbits, with some songs going as low as 145 and others as high as 215. The only drawback is that the encoding speed is quite slow compared to iTunes.

Originally posted by ast3r3x (in regard to der Kopf's comments about lossiness when converting MP3 to AAC)

lossy compared to what wav, midi, shorten?

Lossy compared to the first compressed file you started with... regardless of the compressed formats involved. If you take audio in any lossy format, and convert it to another lossy format, you compound the signal distortion that's caused by lossy compression.

You can't improve and already-compressed audio file by converting into another format. At best, you break even -- MP3 -> AIFF, for example. Taking a 224Kbps MP3 file and converting it into a 224Kbps AAC file can only make it sound worse. The potential for 224K AAC to sound better than 224K MP3 will only be realized if you go back to the original uncompressed source before you create a new AAC file.

This is not to say their might not be practical reasons to convert between formats, or that a loss in quality caused by such a conversion can't be an acceptable trade-off in some situations. But you'd be missing the boat if you thought that converting to one format from another, without going back to original sources, would be a way to improve sound quality.

Lossy compared to the first compressed file you started with... regardless of the compressed formats involved. If you take audio in any lossy format, and convert it to another lossy format, you compound the signal distortion that's caused by lossy compression.

You can't improve and already-compressed audio file by converting into another format. At best, you break even -- MP3 -> AIFF, for example. Taking a 224Kbps MP3 file and converting it into a 224Kbps AAC file can only make it sound worse. The potential for 224K AAC to sound better than 224K MP3 will only be realized if you go back to the original uncompressed source before you create a new AAC file.

This is not to say their might not be practical reasons to convert between formats, or that a loss in quality caused by such a conversion can't be an acceptable trade-off in some situations. But you'd be missing the boat if you thought that converting to one format from another, without going back to original sources, would be a way to improve sound quality.